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检索条件"机构=Department for Design Problems in Microelectronics"

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Methods for Reducing the Number of Matching Operations in the Content-Addressable Memory of the Parallel Dataflow Computing System 'Buran' 3

Methods for Reducing the Number of Matching Operations in th...

引用

3rd IEEE International Conference on problems of Informatics, Electronics and Radio Engineering, PIERE 2024

作者： Levchenko, Nickolay N. Zmejev, Dmitry N. Klimov, Arkadiy V. Nrc 'Kurchatov institute' Department for Design Problems in Microelectronics Moscow Russia

ISBN: (纸本)9798331516321

The paper describes the dataflow computing model and the architecture of the parallel dataflow computing system implementing it. This computing model and architecture are one of the approaches to solving the problem of increasing the real performance of computing systems. The paper describes the principles of operation of hardware content-addressable memory, which most effectively implements the principles of the dataflow computing model. One of the important problems with such memory is its high power consumption during matching operations. Methods for reducing the number of matching operations and the number of compared bits (bits involved in the matching process) are proposed. Methods for reducing the number of matching operations aim to reduce 'parasitic' comparisons of bits and keys in the token matching process. The studies conducted on the behavioral block-register model of the system and the emulator of the system confirmed the applicability of these methods to reduce the number of bits compared when executing dataflow programs. © 2024 IEEE.

关键词： content-addressable memory dataflow computing model matching operations parallel dataflow computing system parasitic matchings

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Signal Source Angular Location Tracking Algorithm 26

Signal Source Angular Location Tracking Algorithm

引用

26th International Conference on Digital Signal Processing and its Applications, DSPA 2024

作者： Djigan, Victor I. Institute for Design Problems in Microelectronics of Ras Department of Integrated Circuits Design Methodology Moscow Russia

ISBN: (纸本)9798350361728

This paper presents an algorithm of the signal source angular location tracking. The tracking is carried out by means of an antenna array. Traditional angular tracking is based on the usage of a monopulse (sum and difference) architecture which combines the output signals of the directional antennas (with mechanical or electronic steering). This requires the usage of a few antennas and a combiner of the antenna output signals. The algorithm, presented in this paper, allows to find the signal source angular location using only one directional antenna that is the antenna array. It is assumed the usage of the antenna array with the digital beamforming. However, the analog phased array can be used as well. This is because the algorithm is based on the processing of the antenna array output signal estimates after the changing (perturbations) only the phases of the array channel signals. This changing can be done by means of the phased shifters in a phased array or by means of the complex-valued weights with the unity absolute values in an antenna array with the digital beamforming. The paper considers the usage of the algorithm in a linear antenna array. To improve the signal-to-noise ratio, the array output samples are processed by a correlator or a matched filter. For this reason, the measured signal must be a pseudo noise sequence. The paper describes the proposed algorithm and the simulation results that confirm the algorithm's operation and demonstrate its efficiency. © 2024 IEEE.

关键词： Beamforming

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Adaptive Processing of Matched Filter Output Signals in Antenna Array 26

Adaptive Processing of Matched Filter Output Signals in Ante...

引用

26th International Conference on Digital Signal Processing and its Applications, DSPA 2024

作者： Djigan, Victor I. Institute for Design Problems in Microelectronics of Ras Department of Integrated Circuits Design Methodology Moscow Russia

ISBN: (纸本)9798350361728

An Adaptive Antenna Array (AAA) is presented in this paper. The least squares criterion adaptive algorithms are used for the array weights calculation. The informational signal of the AAA is a periodic pseudo random sequence, which is hidden by the noise of the array channels. Such sequences are often used in modern radar, navigation and communication systems. The Matched Filters (MF) or the correlators are used for the processing of the AAA channel signals and this enables the usage of the adaptive algorithms based on the least squares criterion. The output signals of the MF are the cross-correlation functions of the AAA channel signals and the pseudo random sequence. The AAA weights calculation algorithm processes the periodic samples of the maximal values of the MF output signals and processes the error signal between the output and desired signals of the array. The desired signal is also composed using the samples of the maximal values of the auto-correlation function of the pseudo random sequence. The calculation of the AAA weights is provided by the Matrix Inversion Lemma Recursive Least Squares algorithm and its two computationally efficient modifications. The simulation demonstrates that the AAA can suppress the interfering signals and can track the informational signal source angular location simultaneously even if the location is initially unknown. This allows to use the AAA for the signal receiving from a moving source in the presence of the interfering signals. © 2024 IEEE.

关键词： Matched filters

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Algorithm of Signal Source Angular Location Searching and Tracking by Two-Dimensional Rectangular Array

Algorithm of Signal Source Angular Location Searching and Tr...

引用

2024 Wave Electronics and its Application in Information and Telecommunication Systems, WECONF 2024

作者： Djigan, V.I. Institute For Design Problems In Microelectronics Of Ras Department Of Integrated Circuits Design Methodology Moscow Russia

ISBN: (纸本)9798350364989

An algorithm of searching and tracking the angular location of a signal source is presented. It is designed for the usage in a two-dimensional equidistant rectangular receiving antenna array. This algorithm uses symmetric perturbations of the array phase distribution on a fixed phase value and uses estimations of the signal values at the array output after these perturbations. It is required to make four such the perturbations in a two-dimensional array or to make two such the perturbations in a one-dimensional linear array. The array has to be a symmetric one for the algorithm usage. This means, that the phase distribution over the array aperture has to be an odd symmetric relatively the array phase center. Because of the phase perturbation, the algorithm can be used in the analog arrays with the phase only control or in the digital beamforming arrays with the complex-valued weights control. To improve the algorithm accuracy, it is suggested to use a correlator or a matched filter to estimate the array output signals. The estimation can be used if the array receives a pseudo random signal. Such the signals are widely used as the informational ones in a number of modern radio systems. Mathematical details of the proposed algorithm are presented. Simulation demonstrates the performance and the efficiency of this algorithm. © 2024 IEEE.

关键词： Matched filters

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Measuring Far-Field Radiation Patterns of Digital Beamforming Antenna Arrays

Measuring Far-Field Radiation Patterns of Digital Beamformin...

引用

2024 Systems of Signals Generating and Processing in the Field of on Board Communications, SOSG 2024

作者： Djigan, V.I. Department of Integrated Circuits Design Methodology Institute for Design Problems in Microelectronics of RAS Moscow Russia

ISBN: (纸本)9798350383614

A method of the antenna array far-field Radiation Pattern (RP) measurement is considered. This method is designed for the usage with the receiving Digital Beamforming Antenna Arrays (DBAA). Such type of antenna arrays typically does not have a radiofrequency output. Hence, the standard methods of the RP measurement, based on the radiation and the reception of the monochromatic probing signal, cannot be used for the DBAA RP measurement. The proposed method assumes the radiation of a probing signal modulated by a pseudo-noise sequence and the base-band processing of the DBAA output signal by means of a matched filter or a correlator. The paper presents the description of the measurement setup and the mathematical explanation of the measuring procedure. It has been shown that the accuracy of the RP measurement depends on the Signal-to-Noise Ratio (SNR) at the outputs of the DBAA channel receivers, the number of the array channels and the number of the pseudo-noise sequence symbols. The accuracy increases on 3 dB each time as the number of the array antennas or the pseudo-noise sequence symbols is doubled. This result is confirmed by the simulation of the RP measurement of the eight and the sixteen channel linear antenna arrays with the usage of the maximum length pseudo-noise sequence of 1023 and 2047 symbols. ©2024 IEEE.

关键词： Beamforming

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Decision Directed Adaptive Antenna Array for Operation in Low SNR Conditions

Decision Directed Adaptive Antenna Array for Operation in Lo...

引用

2024 Systems of Signal Synchronization, Generating and Processing in Telecommunications, SYNCHROINFO 2024

作者： Djigan, V.I. Institute for Design Problems in Microelectronics of Ras Moscow Department of Integrated Circuits Design Methodology Moscow Russia

ISBN: (纸本)9798350373448

This paper presents an Adaptive Antenna Array (AAA) which is able to suppress the signals of the interferences at its output, to receive the informational signal under the low Signal-to-Noise Ratio (SNR) conditions and to steer its beam (the radiation pattern main lobe) towards the source of the informational signal even if the angular location of the source is initially unknown. It is presented the AAA architecture. The architecture is compared with other architectures of the AAA that are able or not able to operate under the same conditions. The ability to operate in low SNR condition is ensured because of the preprocessing of the AAA channel signals by the correlators or by the matched filters. Such preprocessing is possible if the received informational signal is a Pseudo Random Sequence (PRS). Using the signal as a desired one also allows to use the least squares criterion adaptive algorithms for the AAA weights calculation. This ensures the interferences suppression and the AAA beam steering simultaneously. The considered AAA is able to process the modulated PRS. This ability is provided by the usage of a slicer (like in a decision directed equalizer) which produces the desired signal from the AAA output signal during the steady-state operation. Two versions of the algorithms for the AAA weights calculations are presented. They are the Recursive Least Squares (RLS) adaptive filtering algorithms based on the inverse QR decomposition. Simulation results demonstrate the performance and the efficiency of the considered AAA. © 2024 IEEE.

关键词： Signal to noise ratio

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Adaptive Arrays for GNSS Receivers 3

Adaptive Arrays for GNSS Receivers

引用

3rd IEEE International Conference on problems of Informatics, Electronics and Radio Engineering, PIERE 2024

作者： Djigan, Victor I. National Research Center 'Kurchatov Institute' Department of Design Problems in Microelectronics Moscow Russia

ISBN: (纸本)9798331516321

An adaptive antenna array of the land-based equipment of the Global Navigation Satellite System has to be able to accept the signals from a several satellites and to suppress a several interfering signals simultaneously. The side lobe canceller is one of the widely used adaptive arrays to do this job. Unfortunately, this array does not ensure any specified value of its radiation pattern towards any direction except the directions towards the interfering signal sources. As a result, this array does not ensure the same conditions for the reception of the signals from all visible satellites. The problem solution, which is proposed in this paper, is to use the linearly constrained array instead of the side lobe canceller. A single constraint in a form of a fixed radiation pattern value towards the normal to the array aperture (zenith direction) allows to produce a beam which is always steered towards this direction in spite of the radiation pattern changing caused by the interfering signals cancellation. This ensures a stable reception of the signals at least from the satellites whose angular locations are within the array beam. The performance of a circular linearly constrained array and a circular side lobe canceller of the nine omnidirectional antennas is compared by means of simulation. A better performance of the circular linearly constrained array comparing to the circular linearly constrained array is demonstrated. This is ensured by a wider beam with a bigger gain of the linearly constrained array. This beam allows to receive the signals from a larger number of visible satellites. These results suggest a preferable usage of the linearly constrained arrays against the side lobe cancellers in the receivers of the Global Navigation Satellite System. © 2024 IEEE.

关键词： Omnidirectional antennas

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Simplified Fast Affine Projection Algorithm in Echo Cancellation Tasks

Simplified Fast Affine Projection Algorithm in Echo Cancella...

引用

2023 Wave Electronics and its Application in Information and Telecommunication Systems, WECONF 2023

作者： Djigan, V.I. Institute for Design Problems in Microelectronics of RAS Department of Integrated Circuits Design Methodology Moscow Russia

ISBN: (纸本)9798350348293

This paper considers an adaptive echo canceller which can be used in tasks of acoustic or electrical echo cancellation. The solution of these tasks usually requires the usage of the adaptive filters with a large number of weights. This means that a lot of computational resources of hardware or software are required for the echo canceller implementation. The usage of the Fast (computationally efficient) Affine Projection (FAP) adaptive filtering algorithm ensures the adaptive filter functional efficiency and complexity tradeoff in the echo cancellation tasks. The algorithm consists of two groups of calculations: a fast calculation of the adaptive filter error and a fast calculation of the normalizing matrix and its inversion. The second group is based on the usage of some calculations adopted from the sliding window Fast Recursive Least Squares adaptive filtering algorithms. However these calculations are potentially unstable. The given paper presents a simplified stable version of the multichannel FAP algorithm. The simplification is achieved due to the diagonalization of the normalizing matrix that ensures a simple procedure of direct calculation of this matrix and its inversion. If the echo signal is noisy (and that what always happens in our real world) then the original and the simplified FAP algorithms demonstrate about the same performance. The simulation of the two-channel acoustic echo cancellers confirms this statement. © 2023 IEEE.

关键词： Adaptive filters

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Adaptive Cancellation of Engine Noise in Vehicle Cabin

Adaptive Cancellation of Engine Noise in Vehicle Cabin

引用

2023 Systems of Signal Synchronization, Generating and Processing in Telecommunications, SYNCHROINFO 2023

作者： Djigan, V.I. Institute for Design Problems in Microelectronics of Ras Department of Integrated Circuits Design Methodology Moscow Russia

ISBN: (纸本)9798350348316

This paper presents the adaptive noise canceller for the suppression of noise caused by a rotated engine in a vehicle cabin. The canceller uses two microphones to accept the background signals for the processing. One microphone is installed close to the vehicle engine, i.e. outside the cabin. It allows to isolate this microphone from the near-end speaker's voice which is accepted by the second microphone and is transmitted to the far-end speaker. At the same time, the engine noise penetrates to the cabin and is accepted by the second microphone. As a result, the transmitted voice signal becomes noisy. This noise has to be cancelled priory the voice signal transmitting. Because the engine noise is rather colored than white, the efficiency of usage in the noise canceller the computationally simple adaptive filters based the gradient search is low. This paper considers the usage of the computationally complex but functionally efficient Recursive Least Squares (RLS) adaptive filters in this task. Simulation demonstrates a better performance of the RLS noise canceller comparing to the gradient search based Normalized Least Mean Square (NLMS) noise canceller in the considered scenario. © 2023 IEEE.

关键词： Adaptive filters

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Multichannel Acoustic Echo Canceller Based on Fast Affine Projection Algorithm 25

Multichannel Acoustic Echo Canceller Based on Fast Affine Pr...

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25th International Conference on Digital Signal Processing and its Applications, DSPA 2023

作者： Djigan, Victor Department of Integrated Circuits Design Methodology Institute for Design Problems in Microelectronics of Ras Moscow Russia

ISBN: (纸本)9798350337235

The task of the identification of the multichannel linear system with long channel impulse responses is considered in this paper. The multichannel acoustic echo cancellation in the closed rooms is an example of this task. Dependently on the room size, its decoration and the signal sampling frequency value the duration of the identified impulse responses may take a few thousand samples. To solve this task a multichannel adaptive filter with the corresponding large number of weights is required. The implementation of such filters is a difficult task as the arithmetic complexity of the adaptive filters depends on the number of the filter weights. Due to this reason the effective time-domain Recursive Least Squares (RLS) adaptive filters even in their fast (computationally efficient) forms are rarely used in the acoustic echo cancellers. The computationally simple but less effective time-domain Least Means Squares (LMS) or the Normalized LMS (NLMS) adaptive filters are usually used for this purpose. However, such filters have a number of well-known drawbacks. The slow convergence is one of them when the correlated or nonstationary signals are processed. Speech is an example of such signals. The frequency-domain LMS/NLMS adaptive filters de-correlate the signals. These filters have a smaller computational complexity comparing to their time-domain prototypes. However, as the signals are processed using the blocks of the samples, the frequency-domain adaptive filters introduce the output signal delay and demonstrate a slower convergence because their weights are updated only once per block. This decreases the tracking ability of the frequency-domain adaptive filters. The ability is an important feature of the adaptive filters in the acoustic echo cancellation task because the acoustic medium is usually a not time-invariant one. The usage of the Fast Affine Projection (FAP) adaptive filter is the tradeoff between the acoustic echo canceller efficiency and its complexity. Su

关键词： Adaptive filters

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